Portech MV-374G-3G GSM/UMTS/VoIP Gateway, 4x SIM, 1x LAN (Global Version: 3G 2G+3G 800/850/900/1900/2100MHZ)

G = Global Version

Artikelnummer	0877
Kategorie	GSM, 3G, VoLTE, LTE-Data
Hersteller

653,31 € (Netto 549,00 €)

inkl. 19% USt. , zzgl.

Sofort verfügbar
Lieferzeit: 2 - 3 Wochen (DE - Ausland abweichend)

4 Channels VoIP GSM/WCDMA/UMTS Gateway
MV-374 is a 4 channels VoIP GSM/WCDMA/UMTS Gateway for call termination (VoIP to GSM/WCDMA/UMTS ) and origination (GSM/WCDMA/UMTS to VoIP). It is SIP based and compatible with Asterisk,SIP Proxy Server,VoipBuster. It can enable to make 4 calls simultaneously from IP phones to GSM/WCDMA/UMTS networks and GSM/WCDMA/UMTS networks to IP phone.

MV-374/MV-378: New version. Build in Dial peer Server,Stun Server and NAT Function.
5060 is behalf no
It has 4/8 lines: 5064,5066,5068,5070,5072,5074,5076,5078
The call automatically switches from a busy line to available line.
So user just send call to 5060 port from Asterisk/IP PBX *5060 Port can be changed to any port user needs

New Developping: STDP Box is released in the market

*It's an external box with Dial Peer Server,Stun Server,NAT function all in one that can handle all MV-37X,total volume up to 8 ports each.
(E.g you can handle MV-370*2,MV-372*3 with one STDP Box)

Option SBK-32 :32 SIMs Remote SIM Bank and SIM Server
Connect with PORTech GSM Gateway via internet
SIM cards no longer need to be installed in GSM Gateway anymore;
You can deploy your GSM Gateway in different locations.
Centralize and supervise all SIMs in one place.

Option GSM Booster: BT-918/BT-921
It can improve the cellular signal, simple installation

Work with Dial peer Server (free)

Dial Peer Server can manage 128 GSM ports at same time
User just need to set one SIP trunk
Send call to dial peer IP:5060 port from Asterisk/IP PBX
The call automatically switches from a busy line to available line.
Provide CDR
Monitor the signal of all GSM ports

Major Function

VoIP(SIP),GSM conversion.
VoIP(SIP),UMTS conversion for all world
50 sets of LAN --> MOBILE routes setting,50 sets of MOBILE --> LAN routes setting.
-Support one stage diaing
-Support free mode-two stage dialing and assigned mode-one stage dialing
Voice response for setting and status(dial in from mobile).
For call termination (VoIP to GSM/UMTS ) and origination ( GSM/UMTS to VoIP).
Standard SIP(RFC2543,RFC3261) protocol,Communicates with other gateway or PC
Receive SMS and Send SMS (2G/3G)
Allows your program Send/receive SMS with AT Command
Call Back feature
All functions can be set on web.
Provide CDR
1 year warranty

Specification

Protocols:SIP (RFC2543,RFC3261)
TCP/IP:IP/TCP/UDP/RTP/RTCP/,CMP/ARP/RARP/SNTP,DHCP/DNS
Client,IEEE802.1P/Q,ToS/DiffServ,NAT Traversal,STUN,uPnP,IP Assignment,Static IP,DHCP,PPPoE
Codec:G.711 u-Law,G.711 a-Law,G.729A,G.729A/B
Voice Quality,VAD,CNG,AEC,LEC,Packet loss

3G Frequecny: